similar to: Weird issues with TDM400P

List, Okay, here's one that has me stumped, and it might just be something simple. Currently, we are setup so that when someone calls in and tries to reach the operator / front desk, it rings several different phones in sequence. (i.e. it rings the front desk for 15 seconds, then a guy down the hall from it for 15 seconds, then my desk for 15 seconds, and as a last resort, my cordless

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

I have been getting that message also. I have been using various versions of CVS head since Feb. 2005. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Hello, I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings, PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet * is for AA / Voicemail and VOIP in/out Currently the AA / Voicemail function for incoming PSTN calls are working well. My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem. But

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am running into some problems here with PrivacyManager. We used to use it without any issue, but now there seems to be several problems. We are currently running Asterisk 1.2.4. First, it seems that if the user does not press the pound (#) key after entering their number, PrivacyManager will fail. I have the minlength set to 10, and

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom

Hi all, I have an existing panasonic analog pbx in use and a asterisk server with digium tdm400p(2 fxs and 2 fxo). channel 1 -> fxs -> telephone channel 2 -> fxs -> telephone channel 3 -> fxo -> extension 15 at panasonic pbx channel 4 -> fxo -> phone line from telco We call in to fxo (channel 4) and enter the ivr which prompt us to enter the extension number. After

Hi, after finally getting my testsystem to work well I'm planning to buy some hardware. Now I was wondering what the advantages / disadvantages of using a Wildcard TDM400P solution over using real sip phones like the snoms is? The only thing I came up with is that the Snoms can use a LDAP adressbook on the server. What about calling a sip adress like person@server.com. Can this be done by

Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P &amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting

Hi all, I am running samba-server-3.0.6-4.1.100mdk, openldap-servers-2.1.25-6mdk, lib64sasl2-plug-digestmd5-2.1.15-10.1.100mdk. I have searched through the lists and I am wondering if I am the only one doing this kind of set-up.. Anyway question is as follows: In my ldap server I have normal posix accounts with plain text password that are sorted out by a sasl-regex in the slapd.conf and

OK, I''ve been fritzing around with something I noticed last night, thinking that I understood what was going on, but now it''s getting confusing again. A system that has been running for a couple of months had a hole in the probe ID list near the end in the middle of the fbt probes. And then a couple of syscall probes were stuck in the hole. It looked like this: ... 40311

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,

hi, I am on fax-to-email system, basically taken from http://www.voip-info.org/wiki-Asterisk+Fax+to+email well, it seems that asterisk causes some problems so that it is impossible to send any fax. I.e. when I try to send a fax to asterisk (rxfax application): normalFax--->FXS(TDM400P)--->Asterisk(RxFax) it breakes up the trasmission at the beginning of sending page. The code given by

Hi everyone, I'd like to try to use * & ASTCC to create a pre-paid public call card solution. In testing is looks good. I've been talking with Telecom suppliers about supplying me with E1 Primary ISDN lines (probably 4 to start with) and I'd purchase digium TE405Ps to connect everything up. What I'm uncertain about is how to handle CPC with asterisk. I know that 2mbit E1

Someone on the list a while back suggested that if you were having problems with call disconnects, to look into a product from Viking TellecomSolutions called cpc-disconnect: http://www.vikingtelecomsolutions.com/catalog/model_CPC-1.htm I received my unit on Friday and put it into place Saturday afternoon (SBC in this area doesn't supply call disconnect supervision). The unit was acting

Hi Guys, I'm looking for a light on the next problem integrating an Asterisk server with a Panasonic PBX using a PRI. The follow config has been working for two weeks and works almost fine, except for some little problems... 1- PSTN provides the E1/PRI (EuroISDN) 2- Asterisk receives the PRI 3- Asterisk provides a second PRI to Panasonic 4- Panasonic receives the asterisk PRI People can

Hello list. I'm going te explain my trouble. I have my asterisk with a TDM400P with 4 FXS channels. Two ports are connected to a Panasonic PBX (it's working fine), and others two ports are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected to a CO port (where i had a pstn line). When I call to the Alcatel PBX, the asterisk show me in it console that es ringing but

The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF integration feature that can be enabled for dedicated voice mail ports. What I want to do is connect an X100P FXO port to a jack on the Panasonic and make use of the Panasonic's DTMF call progress tones that it provides in DTMF integration mode. The integration works well when a Panasonic extension is forwarding into